265a5d298a
BUG=314 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1385004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3933 4adac7df-926f-26a2-2b94-8c16560cd09d
53 lines
1.5 KiB
C++
53 lines
1.5 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
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#define WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
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#include "video_coding.h"
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#include "module_common_types.h"
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#include "common_types.h"
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#include "rtp_rtcp.h"
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#include "typedefs.h"
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#include "test_util.h"
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#include <string>
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#include <stdio.h>
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class RtpDataCallback : public webrtc::RtpData {
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public:
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RtpDataCallback(webrtc::VideoCodingModule* vcm) : vcm_(vcm) {}
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virtual ~RtpDataCallback() {}
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virtual int32_t OnReceivedPayloadData(
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const uint8_t* payload_data,
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const uint16_t payload_size,
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const webrtc::WebRtcRTPHeader* rtp_header) {
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return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
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}
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private:
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webrtc::VideoCodingModule* vcm_;
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};
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int RtpPlay(const CmdArgs& args);
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int RtpPlayMT(const CmdArgs& args);
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int ReceiverTimingTests(CmdArgs& args);
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int JitterBufferTest(CmdArgs& args);
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int DecodeFromStorageTest(const CmdArgs& args);
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// Thread functions:
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bool ProcessingThread(void* obj);
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bool RtpReaderThread(void* obj);
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bool DecodeThread(void* obj);
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bool NackThread(void* obj);
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#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
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